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 Duration 21 hours

Course Outline

Part I: Introduction

  1. Introduction
    1. Background and rationale for adoption
    2. Classification of VoIP technologies and their evolution
    3. SIP – core principles and concepts
    4. SIP standardization frameworks (RFC 3261 and associated industry standards)
  2. System Architecture
    1. UA – User Agent components
    2. Standardized server roles: Registrar, Location, Proxy, and Redirect
    3. Application server integration
    4. Identity verification and addressing mechanisms
    5. The SIP architectural model
  3. Server Operations and Functionality
    1. Registration processes
    2. Server behavior in Proxy and Redirect modes
    3. Differences between stateless and stateful Proxy implementations
    4. Location server functionalities
    5. SRV records and DNS resolution
    6. uri/url/urn structures, ENUM, and NAPTR records
  4. SIP Signaling Protocols (including Instant Messaging & Presence – IMP extensions)
    1. Message structure and format
    2. Request types and handling
    3. Response types and processing
    4. Illustrative call cycle analysis
    5. Header definitions and parameter usage
    6. IMP operational models
  5. SDP (Session Description Protocol)
    1. Media description methods
    2. Standardized codec lists
    3. Rules for session negotiation
  6. Call Flows and SIP Signaling
    1. SIP session establishment – Reference to RFC 3261
    2. Detailed analysis of sample call scenarios
    3. Conferencing protocols and IP PBX integration
    4. Dynamics of changing media within a session
    5. Implementation of IMP features
  7. Routing of SIP Requests and Responses
    1. VIA header processing
    2. ROUTE and RECORD-ROUTE header functions
  8. Interoperability with PSTN
    1. SIP-T and SIP-I standards
    2. SIP early media and trunking configurations
    3. Signaling alignment between SIP and PSTN
  9. SIP Security Considerations
    1. Secure SIP, Secure RTP, and Secure RTCP protocols
    2. Common implementation strategies for Secure SIP
  10. Operational Challenges and Future Directions
    1. Traversal techniques for NAT and firewalls
    2. Quality of Service (QoS) management
    3. Integration of SIP and SDP within 3GPP IMS architectures
  11. Summary and Strategic Discussion

Part II: Practical Application

  1. SIP Implementation in LAN Environments: XLite SIP UA + Asterisk
    1. Establishing Asterisk accounts with basic dial plans
    2. Configuring XLite SIP UA (dtmf, codecs, nat, rtp, timer, register) and various SIP endpoints (Polycom, Gigaset, Yealink, Linphone)
    3. Processes for registration, call initiation, and reception
    4. Peer-to-peer (P2P) call testing with Linphone
  2. Analysis of SIP Signaling Using Wireshark
    1. Server configuration for monitoring
    2. Tracking SIP signaling and RTP media streams
    3. Packet-level analysis and specific call isolation
    4. Voice quality assessment: Jitter buffers, DTMF signaling retrieval (RFC 2833, INFO), and troubleshooting codecs and DTMF (transcoding, GSM issues, tone duplication)
    5. Integration of VoIP monitoring tools
  3. SDP, Instant Messaging, and Presence (IM&P)
    1. Review of SDP parameters and attributes
    2. Application of SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
    3. Practical exercises for IM&P using XLite and Linphone
  4. SIP Call Flow Dynamics
    1. SIP Registration via DNS
      • Analysis of SIP SRV records
      • Endpoint registration utilizing DNS-SRV
    2. Call Flows involving DNS
      • Signaling analysis using Wireshark
      • Diagnosing DNS timeouts and latency issues
  5. Managing SIP Trunks
    1. Setup of test SIP trunk connections
    2. Mitigation of threats (DOS, DDOS, fraud, CPS limits)
  6. SIP Security Posture
    1. Implementing SIP security with IPSec
    2. Enhancing security via Secure SIP protocols
    3. Assessing fraud risks in IP telephony
    4. Defense against DDOS and other attack vectors
  7. Deployment of SIP-Based VoIP Services
    1. Switch configuration standards
    2. SIP client setup and registration procedures
    3. Software Stack
      • Asterisk PBX / Freeswitch softswitch / Cisco Call Manager
      • Linux CentOS operating system
      • TDM2IP drivers
      • Softphone applications (XLite, Linphone)
    4. Hardware Infrastructure
      • Server systems
      • TDM2IP cards/gateways
      • Hardphone devices (Polycom, Gigaset, Yealink)
  8. Endpoint Configuration: Softphones and Hardphones
    1. Configuration Settings
      • Codec selection
      • Credential and server parameters (User/Password/SIP Server/Proxy/Ports)
    2. Operation and Signaling for:
      • Three-way calling capabilities
      • Call forwarding mechanisms
      • Attendant call transfer
      • Message Waiting Indication (MWI) and Busy Lamp Field (BLF)
    3. Yealink autoprovisioning protocols
    4. Vendor-specific operational constraints
  9. SIP and Network Address Translation (NAT) Challenges
    1. NAT classification and structural analysis
    2. Application of STUN (Simple Traversal of UDP Through NATs)
  10. VoIP Call Quality Assurance and Troubleshooting
    1. Diagnosing connected calls with missing media
    2. Critical QoS indicators
      • Latency, jitter, and play buffer sizing
    3. Voice quality metrics
      • RTCP analysis for delay and jitter
      • MOS calculation per ITU-T G.107 E-model
    4. Utilization of VoIP quality monitoring tools (Voipmonitor)
  11. Cloud-Hosted IP Telephony Solutions
  12. Conclusion and Addressing Participant-Submitted SIP and VoIP Issues

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